Heretofore, oversampling processing to convert a sampling frequency to a value several times higher than the original value is performed before a digital audio signal is input to a digital/analog converter. With this arrangement, the phase feature of an analog anti-aliasing filter keeps the digital audio signal outputted from the digital/analog converter, at a constant level in the audible high frequency band, and prevents influences of digital image noises caused by sampling.
Typical oversampling processing employs a digital filter of the primary linear (straight line) interpolation system. Such digital filter is used for creating linear interpolation data by averaging plural pieces of existing data when the sampling rate is changed or data is missing.
Although the digital audio signal subjected to the oversampling processing has an amount of data several times more than that of the original data in the direction of time-axis because of linear interpolation, the frequency band of the digital audio signal subjected to the oversampling processing is not changed so much and the sound quality is not improved as compared with before. Moreover, since the data interpolated is not necessarily created based on the waveform of the analog audio signal before it is A/D converted, the waveform reproducibility is not improved at all.
Furthermore, in the case of dubbing digital audio signals having different sampling frequencies, the frequencies are converted by means of the sampling rate converter. In such cases, however, the linear digital filter can interpolate only linear data, so that it is difficult to improve the sound quality and waveform reproducibility. Furthermore, in the case where data samples of digital audio signal are missing, the same results as those of the above occurs.